October 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by Sebastian Schneider 08 Oct '08

08 Oct '08

What about we start porting the driver on our own? -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 07 Oct '08

07 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f5a6dbb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,10 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_SND_SOC_DEBUG_FS + struct dentry *debugfs_root; + struct dentry *debugfs_codec_reg; +#endif }; /* runtime channel data */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..aad7e0b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,14 @@ if SND_SOC config SND_SOC_AC97_BUS bool +config SND_SOC_DEBUG_FS + boolean "changing codec registers through debugfs (DEVELOPMENT)" + depends on DEBUG_FS + help + The ability to directly write codec registers through + /sys/kernel/debug/xx/codec_reg is useful when developing + a new codec. + # All the supported Soc's source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..46827cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,113 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_SND_SOC_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t res; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + + res = soc_codec_reg_show(devdata, buf); + res = simple_read_from_buffer(user_buf, count, ppos, buf, res); + kfree(buf); + return res; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + socdev->debugfs_codec_reg = debugfs_codec_reg; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove(socdev->debugfs_codec_reg); + debugfs_remove(socdev->debugfs_root); + socdev->debugfs_codec_reg = NULL; + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1320,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1254,6 +1358,7 @@ free_card: if (codec->card) snd_card_free(codec->card); device_remove_file(socdev->dev, &dev_attr_codec_reg); + soc_cleanup_debugfs(socdev); mutex_unlock(&codec->mutex); } EXPORT_SYMBOL_GPL(snd_soc_free_pcms); -- 1.5.4.3

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[alsa-devel] [PATCH 1/1] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 07 Oct '08

07 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..1077e9d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,8 +3016,8 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); @@ -3031,20 +3031,20 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &ua700_ep }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { + static const struct snd_usb_midi_endpoint_info uaxx_ep = { .out_cables = 0x0001, .in_cables = 0x0001 }; - static const struct snd_usb_audio_quirk ua25_quirk = { + static const struct snd_usb_audio_quirk uaxx_quirk = { .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep + .data = &uaxx_ep }; if (chip->usb_id == USB_ID(0x0582, 0x002b)) return snd_usb_create_midi_interface(chip, iface, &ua700_quirk); else return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); + &uaxx_quirk); } if (altsd->bNumEndpoints != 1) @@ -3376,9 +3376,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..815bf36 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1197,15 +1197,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1338,6 +1338,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

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Re: [alsa-devel] 2 Au1xxx (Alchemy) AC97 drivers
by Manuel Lauss 07 Oct '08

07 Oct '08

On Tue, Oct 07, 2008 at 08:28:26AM -0500, Kevin Hickey wrote: > There appear to be 2 AC97 drivers for Alchemy in the latest Linux tree. > One is in sound/mips and the other is in sound/soc. Are both going to > be maintained going forward? If not, which one will be? The one in sound/soc is for Au1200/Au1550; I think the sound/mips one only works on Au1000/Au1500. Manuel Lauss -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [topic/asoc][PATCH 0/2] minor clean-ups and aic3x naming change
by Jarkko Nikula 07 Oct '08

07 Oct '08

These two patches are generated agains Takashi's topic/asoc branch. First one will remove unused AUDIO_NAME define from codec drivers. IMO it didn't look so readable to use this define so I end up removing it. Second one will use uniform naming like other tlv320aic drivers for tlv320aic3x. -- Jarkko

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[alsa-devel] size of snd_hda_intel
by Jon Smirl 07 Oct '08

07 Oct '08

>ismod snd_hda_intel 435,772 435,772 bytes seems awfully large for an audio driver. The large size seems to be due to having support for multiple codecs in the driver. Could a generic hda driver load the correct codec dynamically instead of building them all in statically? I have a Realtek codec. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Fwd: AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

---------- Forwarded message ---------- From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> Date: Tue, Oct 7, 2008 at 5:36 PM Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 To: Arun KS <arunks(a)mistralsolutions.com> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" Will that be the reason ???? On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> wrote: > > On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: > > > >> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for the > past > >> few days. I got the driver patch from the linux-omap/alsa-devel mailing > >> lists. The driver got compiled along just fine. But when the kernel > boots, > >> no sound cards have been listed. > > > > CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that > > support. > > > >> Advanced Linux Sound Architecture Driver Version 1.0.17. > > > >> ASoC version > >> 0.13.2 > >> AIC23 Audio Codec > >> 0.1 > >> MCLK = 12000000 [12000000], usecount = > >> 0 > >> ALSA device > >> list: > >> No soundcards > >> found. > > > >> Any one have tested this before ? > > Can you check i2c? > > # cat /sys/bus/i2c/devices/1-001b/name > tlv320aic23 > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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[alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

Hi all, I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for the past few days. I got the driver patch from the linux-omap/alsa-devel mailing lists. The driver got compiled along just fine. But when the kernel boots, no sound cards have been listed. Advanced Linux Sound Architecture Driver Version 1.0.17. ASoC version 0.13.2 AIC23 Audio Codec 0.1 MCLK = 12000000 [12000000], usecount = 0 ALSA device list: No soundcards found. Any one have tested this before ? - Jesslyn Abdul Salam

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[alsa-devel] [PATCH 2.6.27?] ASoC: Correct inverted Mic PGA Switch control in wm8510 driver
by Mark Brown 07 Oct '08

07 Oct '08

Mic PGA Switch should be inverted in the WM8510 driver but isn't. Reported-by: ext-jukka.hynninen(a)vaisala.com Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 75ed041..9a37c8d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -199,7 +199,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; static const struct snd_kcontrol_new wm8510_boost_controls[] = { -SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 1), SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), }; -- 1.5.6.5

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[alsa-devel] [PATCH] snd-core: enlarge snd_card.components for up to 4 codecs
by Wu Fengguang 07 Oct '08

07 Oct '08

Enlarge snd_card.components[80] to 128 bytes, with space for 4 codecs. The previous size 80 cannot support HP 2230s which has 3 codecs. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/include/sound/asound.h b/include/sound/asound.h index ca2f358..b3801b0 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -708,7 +708,7 @@ struct snd_ctl_card_info { unsigned char longname[80]; /* name + info text about soundcard */ unsigned char reserved_[16]; /* reserved for future (was ID of mixer) */ unsigned char mixername[80]; /* visual mixer identification */ - unsigned char components[80]; /* card components / fine identification, delimited with one space (AC97 etc..) */ + unsigned char components[128]; /* card components / fine identification, delimited with one space (AC97 etc..) */ unsigned char reserved[48]; /* reserved for future */ }; diff --git a/include/sound/core.h b/include/sound/core.h index f52ab6f..dcbc424 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -116,7 +116,7 @@ struct snd_card { char shortname[32]; /* short name of this soundcard */ char longname[80]; /* name of this soundcard */ char mixername[80]; /* mixer name */ - char components[80]; /* card components delimited with + char components[128]; /* card components delimited with space */ struct module *module; /* top-level module */

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[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for process_unsol_events(). Signed-off-by: Wu Fengguang<wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..7717a92 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -314,7 +314,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) } /* - * process queueud unsolicited events + * process queued unsolicited events */ static void process_unsol_events(struct work_struct *work) {

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[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for is_in_nid_list(). Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..6268667 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2807,7 +2807,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } /* - * Helper for automatic ping configuration + * Helper for automatic pin configuration */ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: add new param mux to dapm_mux_update_power
by Richard Zhao 07 Oct '08

07 Oct '08

Function dapm_mux_update_power needs enum index mux and register mask value val as parameters, but it only has a parameter val, and uses it as both val and mux. snd_soc_test_bits(widget->codec, e->reg, mask, val) val is register mask here, e->texts[val] but val should be enum index mux here. This patch adds a new param mux to fix it. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: Fix wrong param when call snd_soc_test_bits
by Richard Zhao 06 Oct '08

06 Oct '08

Fix wrong param when call snd_soc_test_bits in function dapm_mux_update_power. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

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Re: [alsa-devel] [Alsa-user] [SOLVED] Re: snd_hda_intel on ALC662 (Classmate 2nd gen) does not mute and does not offer mic boost
by Vedran Miletić 06 Oct '08

06 Oct '08

http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=ac… See diff of that commit for some hints. Or don't. I got it wrong. :-( Basically, you need to find out subsystem ID for your soundcard. "lspci -vnn" will be your friend here. You should find out something like this for your soundcard: 00:07.0 Audio device [0403]: nVidia Corporation Device [10de:0774] (rev a1) Subsystem: ASRock Incorporation Device [1849:3662] <------ THIS IS THE IMPORTANT ROW Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 21 Memory at f9e78000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel So I should have put 0x1849, 0x3662 for my souncard instead of 0x1849, 0x0774. I probably wouldn't notice, but now I will patch it. Thanks for asking! 2008/10/4 Pedro Pessoa <pessoa(a)angulosolido.pt>: > On Saturday 04 October 2008 17:54, Vedran Miletić wrote: >> Can you create a patch to add this device to quirks list? > > Sure! Let me know what I have to do. > > The quirks list is something that will make 'auto' load ECS? > > -- > Angulo Sólido - Tecnologias de Informação > http://angulosolido.pt > -- Vedran Miletić

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[alsa-devel] [PATCH] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 06 Oct '08

06 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic, moving the MIDI interfaces to the quirk data header. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..05c7849 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,37 +3016,12 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); - if (altsd->bNumEndpoints == 2) { - static const struct snd_usb_midi_endpoint_info ua700_ep = { - .out_cables = 0x0003, - .in_cables = 0x0003 - }; - static const struct snd_usb_audio_quirk ua700_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua700_ep - }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk ua25_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep - }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); - } - if (altsd->bNumEndpoints != 1) return -ENXIO; @@ -3376,9 +3351,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..ad7a318 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } }, { .ifnum = -1 @@ -1197,15 +1201,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } }, { .ifnum = -1 @@ -1338,6 +1346,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

4 3

Re: [alsa-devel] ICH9 Sound
by Ronald Horn 06 Oct '08

06 Oct '08

Hello, all mixer levels are ok. I have several times checked all settings. I deinstalled and reinstalled the drivers in yast2, too. The phenomenon is that then a sound is playing, once an incredible noice is playing over the original sound. And this sound is stopping first about ten seconds after I closed the player. It's like an feedback loop. Greetings Ronald > ICH9 sound should be supported. And, as you see, your card is > detected. Perhaps the codec or that specific notebook isn't supported. > Are you sure mixer levels are adjusted properly? > > I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, > > because there is no driver witch works. > > Here are some reports from my system: > > / > > lspci -nv/ > > 00:1b.0 0403: 8086:293e (rev 03) > > Subsystem: 103c:30f4 > > Flags: bus master, fast devsel, latency 0, IRQ 22 > > Memory at df300000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: [50] Power Management version 2 > > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > > Queue=0/0 Enable+ > > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > > Capabilities: [100] Virtual Channel <?> > > Capabilities: [130] Root Complex Link <?> > > Kernel driver in use: HDA Intel > > Kernel modules: snd-hda-intel > > > > / > > cat /proc/asound/cards/ > > 0 [Intel ]: HDA-Intel - HDA Intel > > HDA Intel at 0xdf300000 irq 22 > > > > > > /head -n 1 /proc/asound/card0/codec*/ > > ==> /proc/asound/card0/codec#0 <== > > Codec: IDT 92HD71B7X > > ==> /proc/asound/card0/codec#2 <== > > Codec: Generic 10de ID 6 > > > > > > I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. > > Maybe your project could develop some drivers.

1 0

[alsa-devel] [PATCH]ASoC: Create a common directory for AVR32 and ARM9 atmel boards.
by Sedji Gaouaou 06 Oct '08

06 Oct '08

Create a common directory for AVR32 and ARM9 atmel boards to unify atmel audio DAI. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 2 +- sound/soc/atmel/Kconfig | 43 ++ sound/soc/atmel/Makefile | 15 + sound/soc/atmel/atmel-pcm.c | 484 +++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.h | 86 ++++ sound/soc/atmel/atmel_ssc_dai.c | 803 ++++++++++++++++++++++++++++++++++++++ sound/soc/atmel/atmel_ssc_dai.h | 121 ++++++ sound/soc/atmel/playpaq_wm8510.c | 523 +++++++++++++++++++++++++ sound/soc/atmel/sam9g20_wm8731.c | 329 ++++++++++++++++ 10 files changed, 2406 insertions(+), 3 deletions(-) create mode 100644 sound/soc/atmel/Kconfig create mode 100644 sound/soc/atmel/Makefile create mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/atmel/atmel-pcm.h create mode 100644 sound/soc/atmel/atmel_ssc_dai.c create mode 100644 sound/soc/atmel/atmel_ssc_dai.h create mode 100644 sound/soc/atmel/playpaq_wm8510.c create mode 100644 sound/soc/atmel/sam9g20_wm8731.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda66..615ebf0 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349..4d475c3 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 0000000..170d9da --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the AT91 SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 0000000..f54a7cc --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c new file mode 100644 index 0000000..6d317d2 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.c @@ -0,0 +1,484 @@ +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* PDC register save */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + snd_pcm_period_elapsed(substream); +} + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params */ + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); + return 0; +} + +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + return 0; +} + +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + +static int atmel_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + out: + return ret; +} + +static int atmel_pcm_close(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, +}; + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 atmel_pcm_dmamask = 0xffffffff; + +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &atmel_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +#ifdef CONFIG_PM +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; + + if (!runtime) + return 0; + + prtd = runtime->private_data; + params = prtd->params; + + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, +}; +EXPORT_SYMBOL_GPL(atmel_soc_platform); + +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 0000000..ec9b282 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 0000000..2ed1575 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,803 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} +#if 0 +/* + * Record the SSC system clock rate. + */ +static int atmel_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* + * The only clock supplied to the SSC is the AT91 master clock, + * which is only used if the SSC is generating BCLK and/or + * LRC clocks. + */ + switch (clk_id) { + case ATMEL_SYSCLK_MCK: + atmel_ssc_sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} +#endif +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params,}, + .dai_ops = { + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 0000000..a828746 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino(a)endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c new file mode 100644 index 0000000..7e6560b --- /dev/null +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -0,0 +1,523 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/at32ap700x.h> +#include <mach/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = snd_soc_dai_set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); + + + + /* Make CSB show PLL rate */ + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_bus = 0, + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 0000000..4e191df --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,329 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino(a)endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_machine snd_soc_machine_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .machine = &snd_soc_machine_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

4 16

[alsa-devel] TASCAM US122L
by Fernando Perfumo 06 Oct '08

06 Oct '08

Hello all, following is an extract of kern.log when connecting and disconecting a TASCAM US122L : ( the pcm part of the us122L does not appear in /proc/asound/... but only the midi part. kernel : 2.6.27-rc5-git11, alsa-driver-1.0.18rc3 compiled separately of the kernel sources, alsa-lib-1.0.18rc3 and alsa-utils-1.0.18rc3 ) I'd appreciate some ideas about the segfault in alsactl and the ausence of the pcm functionality.. Thanks many, Fernando. Oct 6 14:18:17 64studio kernel: usb 2-1: new high speed USB device using ehci_hcd and address 10 Oct 6 14:18:17 64studio kernel: usb 2-1: config 1 interface 1 altsetting 1 bulk endpoint 0x83 has invalid maxpacket 9 Oct 6 14:18:17 64studio kernel: usb 2-1: configuration #1 chosen from 1 choice Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:538: ffff880037942000:0 Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:538: ffff88005e018000:1 Oct 6 14:18:17 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:103: 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/us122l.c:103: 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-18 ul=1518 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=0 ul=1554 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:714: 23 106 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 1 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-12 ul=1524 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:61: idle=-30 ul=1518 ps=1536 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 6 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 11 46 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 88 46 -6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 176 46 -12 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 0 253 46 -18 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 6 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 11 46 0 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 1518 242 46 -6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 3036 6 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 4590 5 46 0 Oct 6 14:18:18 .... .............................................................. Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 44538 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 46056 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 50688 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-ker.................................. ......................................................................................................................................................................... Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 93696 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 95220 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 99846 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb.............................................................. ...................................................................................................................................................................... Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 153576 5 47 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 158208 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 165882 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:552: 173556 5 46 6 Oct 6 14:18:18 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:738: 3 Oct 6 14:18:19 64studio last message repeated 2 times Oct 6 14:18:19 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usbmidi.c:1368: created 1 output and 1 input ports Oct 6 14:18:19 64studio kernel: alsactl[25592]: segfault at 0 ip 00007fb5c9e2f5a0 sp 00007fffd26840b8 error 4 in libc-2.3.6.so[7fb5c9dbd000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25594]: segfault at 0 ip 00007fc0e9fc45a0 sp 00007ffff2819238 error 4 in libc-2.3.6.so[7fc0e9f52000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25670]: segfault at 0 ip 00007fd7550975a0 sp 00007fff5d8ee318 error 4 in libc-2.3.6.so[7fd755025000+121000] Oct 6 14:18:19 64studio kernel: alsactl[25671]: segfault at 0 ip 00007f2ab8cc65a0 sp 00007fffc151cf38 error 4 in libc-2.3.6.so[7f2ab8c54000+121000] Oct 6 14:18:47 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:456: underrun, status=4294967225 Oct 6 14:18:47 64studio kernel: ALSA /usr/src/alsa-driver-1.0.18rc3/usb/usx2y/../../alsa-kernel/usb/usx2y/usb_stream.c:456: underrun, status=4294967225 Oct 6 14:18:47 64studio kernel: usb 2-1: USB disconnect, address 10

2 1

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with sys filesystem
by Troy Kisky 06 Oct '08

06 Oct '08

i.e. echo 6 59 >/sys/devices/platform/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..df23b23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1002,7 +1002,33 @@ static ssize_t codec_reg_show(struct device *dev, return count; } -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); + +static ssize_t codec_reg_write(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) +{ + char *start = (char *)buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = dev_get_drvdata(dev); + struct snd_soc_codec *codec = devdata->codec; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return start - buf; +} + +static DEVICE_ATTR(codec_reg, 0644, codec_reg_show, codec_reg_write); /** * snd_soc_new_ac97_codec - initailise AC97 device -- 1.5.4.3

3 3

[alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Klaas De Craemer 06 Oct '08

06 Oct '08

Hello, A few days ago, I posted a patch to the linux kernel list, but afterwards I realised that it is more appropriate for the alsa list. So I repost it here. Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound going to the headphone jack goes comes directly from the DAC, resulting in very low volume output. The patch changes the codec's widget connections so that the internal mixer is put in between and sets its input amplifiers to the maximum value. The output to the internal speaker port has been left untouched. I have tested this and it gives a much better output. Klaas De Craemer diff -uNr linux-2.6.26.5/sound/pci/hda/patch_sigmatel.c linux-2.6.26.5.new/sound/pci/hda/patch_sigmatel.c --- linux-2.6.26.5/sound/pci/hda/patch_sigmatel.c 2008-09-08 19:40:20.000000000 +0200 +++ linux-2.6.26.5.new/sound/pci/hda/patch_sigmatel.c 2008-10-02 20:08:51.000000000 +0200 @@ -624,6 +624,22 @@ {} }; +static struct hda_verb stac92hd71b7_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* connect headphone jack to mixer */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Increase mixer volume for the two dac's */ + { 0x17, 0x370, 0x7f }, + { 0x17, 0x371, 0x7f }, + {} +}; + static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -3642,6 +3658,10 @@ spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; + case 0x111d76b2: /* Dell E6500 stac92hd71b7 configuration */ + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71b7_core_init; + break; default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init;

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[alsa-devel] [patch] bugfix: SOUND: SOC: OMAP: Add support for Gumstix Overo
by David Brownell 06 Oct '08

06 Oct '08

From: David Brownell <dbrownell(a)users.sourceforge.net> Get rid of bogus ASOC boot messages on non-Overo boards, such as with a kernel built for both Overo and Beagle. Signed-off-by: David Brownell <dbrownell(a)users.sourceforge.net> Acked-By: Steve Sakoman <steve(a)sakoman.com> --- I'm not sure where the four patches for Overo support are in the ALSA queue (that's how they'll merge, yes?) but I'll suggest this one merge into the initial "overo.c" submit... sound/soc/omap/overo.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -107,11 +107,11 @@ static int __init overo_soc_init(void) { int ret; - printk(KERN_INFO "overo SoC init\n"); if (!machine_is_overo()) { - printk(KERN_ERR "Not Overo!\n"); + pr_debug("Not Overo!\n"); return -ENODEV; } + printk(KERN_INFO "overo SoC init\n"); overo_snd_device = platform_device_alloc("soc-audio", -1); if (!overo_snd_device) { @@ -135,13 +135,12 @@ err1: return ret; } +module_init(overo_soc_init); static void __exit overo_soc_exit(void) { platform_device_unregister(overo_snd_device); } - -module_init(overo_soc_init); module_exit(overo_soc_exit); MODULE_AUTHOR("Steve Sakoman <steve(a)sakoman.com>");

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[alsa-devel] [PATCH] ALSA: hda: add more board-specific information for Realtek ALC662 rev1
by Vedran Miletić 05 Oct '08

05 Oct '08

I recently got a chance to play with two boards with ALC662 rev1: * BIOSTAR TA780G M2+ * ASROCK K10N78FullHD-hSLI R3.0 Both use 3 stack, 6ch mode with digital out. Since autodetection isn't able to figure that out from BIOS, we need to specify that manually. BTW, thanks for fixing NVIDIA MCP78S [GeForce 8200] HD audio in 1.0.18, it's much appreciated :-) -- Vedran Miletić

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[alsa-devel] [PATCH 2.6.27?] ASoC: Correct inverted Mic PGA Switch control in wm8510 driver
by Mark Brown 04 Oct '08

04 Oct '08

Mic PGA Switch should be inverted in the WM8510 driver but isn't. Reported-by: ext-jukka.hynninen(a)vaisala.com Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This bugfix applies to 2.6.27 but the issue is not very severe - it'd be good to get it in but it's not critical. sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 75ed041..f644da6 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -199,7 +199,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; static const struct snd_kcontrol_new wm8510_boost_controls[] = { -SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Mic PGA Switch", WM8509_INPPGA, 6, 1, 1), SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), }; -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 04 Oct '08

04 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic, moving the MIDI interfaces to the quirk data header. Added a new quirk for the Edirol UA-4FX. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c5cf682..05c7849 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,37 +3016,12 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); - if (altsd->bNumEndpoints == 2) { - static const struct snd_usb_midi_endpoint_info ua700_ep = { - .out_cables = 0x0003, - .in_cables = 0x0003 - }; - static const struct snd_usb_audio_quirk ua700_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua700_ep - }; - static const struct snd_usb_midi_endpoint_info ua25_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk ua25_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_ep - }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &ua25_quirk); - } - if (altsd->bNumEndpoints != 1) return -ENXIO; @@ -3376,9 +3351,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..ad7a318 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } }, { .ifnum = -1 @@ -1197,15 +1201,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } }, { .ifnum = -1 @@ -1338,6 +1346,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

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